Techniques for communicating telephone conversations in digital format have become commonplace in recent years. The telephone signal is usually filtered to limit its bandwidth and is then sampled at a rate that is at least twice the frequency of its highest frequency component. The repetitive samples are applied to an A/D converter to obtain digital representations of the analog samples. Although volume compression and expansion may be utilized to increase the system's dynamic range, it is generally conceded that at least eight bits are required for each digital sample in order that the quantization noise be held to an acceptable minimum. Thus, the generally accepted digital bandwidth required to digitally communicate a telephone signal that is frequency-limited to 4 kilohertz, is at least 2 times 4 times 8, or 64 kilobits per second.
Synchronous time-division multiplexing (TDM) is generally employed to simultaneously carry a plurality of digitized telephone conversations over a single digital channel. With synchronous TDM, each digital telephone channel occupies a precise "time slot" in a high bit-rate serial data sequence. Synchronous TDM accommodates a fixed number of telephone channels and requires that a "time slot" be assigned to each channel whether or not the channel is being used. Such inflexibility can be very wasteful of digital bandwidth. To accommodate 40 unidirectional telephone channels by conventional synchronous TDM would require a digital bandwidth of approximately 40 times 64 kilobits, or 2.56 megabits per second.
It is desirable for economic reasons to increase the digital efficiency of a telephone system so that a large number of telephone channels can be accommodated by a given digital bandwidth. The digital efficiency of conventional TDM can be increased by the use of digital speech interpolation (DSI) techniques. With DSI, a "time slot" is only assigned to a particular telephone channel during periods of actual speech and is dynamically reassigned to another telephone channel during a speech pause. Since pauses are known to occupy about 60 percent of a typical speech pattern, DSI can theoretically increase digital efficiency by about a factor of 2.5. Because of statistical variations in speech patterns, however, a factor of 1.5 is more typically obtained in practice. In addition, DSI efficiency is further reduced by the fact that a "channel assignment table" must be transmitted each TDM cycle to permit the receiver to unambiguously identify the current occupant of each "time slot".
Instead of synchronously transmitting individual digital samples of speech in TDM "time slots", longer sequences of samples can be assembled and transmitted asynchronously as packets--each packet being identified by a packet "header". As with DSI TDM, the digital efficiency of a multiplexed packet telephone system can be increased by a factor of about 1.5 by suppressing speech pauses and transmitting only packets containing samples obtained during spurts of actual speech. Because of the asynchronous nature of packet transmission, however, such systems have required that a "time stamp" be appended to each sequence of actual speech samples so that the appropriate speech pauses could be correctly re-inserted at the receiving terminal. Such "time stamping" increases packet overhead and seriously complicates the speech reconstruction process. An example of a multiplexed speech transmission system employing "time stamped" packets of digitized speech samples has been disclosed in Flanagan U.S. Pat. No. 4,100,377.
Recent advances in technology have provided means for new approaches to increasing the digital efficiency of multiplexed digital telephone systems. In particular, the recent development of high-performance microprocessors capable of many complex calculations per second has made possible the real-time implementation of powerful, but computationally intensive, speech compression algorithms. A variety of such computational algorithms are presently available and are well-known to those of ordinary skill in the art. Such algorithms typically rely upon the principles of time-domain harmonic scaling, transform coding, linear or adaptive predictive coding, sub-band coding, or combinations thereof.
All such computational speech compression algorithms share one common feature. They can effectively transform relatively large groups of digitized speech samples into much smaller sequences, or "frames", of digital compression variables at the transmitting end of a circuit for transmission efficiency, and then expand the frames at the receiving end to approximate the original larger groups of speech samples while still maintaining acceptable speech fidelity. Using techniques that are known to those of ordinary skill in the art, compression ratios--defined as the number of bits in a frame of digital compression variables divided by the number of bits in the corresponding group of digital speech samples--of less than 0.25 can presently be routinely achieved with computational algorithms which still maintain telephone-quality speech. The real-time implementation of such computational speech compression algorithms in a multiplexed digital telephone system has, however, not heretofore been accomplished.